1. Field of the Invention
The invention relates generally to communication network, and more particularly to a system and method for effective transportation of packetized data communication.
2. Discussion of the Background
Communication network includes devices that comprise of hardware and software systems and utilizes interdependent processes to enable the processing and transmission of analog and digital signals. The transmission of signals are done seamlessly across and between circuits switched and packet switched networks. As an example, a voice over packet gateway enables the transmission of human voice from a conventional public switched network to a packet switched network. Such media over packet communication devices (e.g., Media Gateways) require substantial processing power with sophisticated software controls and applications to enable the effective transmission of data from circuit switched to packet switched networks and back again. One form of media transmission, referred to as voice-over-IP (VoIP), is the transport of voice traffic through the use of the Internet protocol.
VoIP requires notably less average bandwidth than a traditional circuit-switched connection for several reasons. First, by detecting when voice activity is present, VoIP can choose to send no data when a speaker on one end of a conversation is silent. Second, the digital audio bit stream utilized by VoIP may be significantly compressed before transmission using a codec (compression/decompression) scheme. However, there can be delay in receiving data packets.
The characteristics of these networks are such that the total delay experienced by each data packet is a function of variable delays due to physical media access and queuing in addition to fixed propagation delays. The result is that the time difference between transmitting any two packets at the source is unlikely to be the same as that observed upon their arrival at the destination. The delay variations are a particular problem for a stream of multimedia packets, because deviations of inter packet delay can have an impact on the audiovisual quality as perceived by the user.
Typically, synchronization methods are used at the receiver to handle delay variations. These synchronization methods usually operate by selectively choosing to drop certain packets deemed to be late or by adding further playback delays to certain packets at the receiver. The playback delay represents delay experienced at the playback location. Thus, the total delay experienced by any packet represents the sum of network delay and playback delay.
The synchronization methods typically store and track the trends of the delays occurring within the network. Since the network characteristics vary with time, these trends vary, and current information is necessary for the methods to be effective. The methods known in the art include the “full aggregation” method wherein all of the data is accumulated into a single distribution curve throughout the lifetime of the transmission or the “flush and refresh” approach in which statistical samples are stored for a period of time and then periodically flushed and refreshed.
However, some of the above methods are not dynamic and are not suitable for a wide range of applications, which are sensitive to total end-to-end delay (TED).
Accordingly, there is a need for a technique that enables an effective and improved method for predicting and adjusting future TED dynamically to measure network delays and thereby, effectively managing packet delays.